/*
  * AAC encoder
  * Copyright (C) 2008 Konstantin Shishkov
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 /**

  * @file

  * AAC encoder
  */
 
 /***********************************
  *              TODOs:

  * add sane pulse detection

  * add temporal noise shaping

  ***********************************/
 

 #include "libavutil/float_dsp.h"

 #include "libavutil/opt.h"

 #include "avcodec.h"

 #include "put_bits.h"

 #include "internal.h"

 #include "mpeg4audio.h"

 #include "kbdwin.h"

 #include "sinewin.h"

 
 #include "aac.h"
 #include "aactab.h"

 #include "aacenc.h"
 
 #include "psymodel.h"

 #define AAC_MAX_CHANNELS 6
 

 #define ERROR_IF(cond, ...) \
     if (cond) { \
         av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
         return AVERROR(EINVAL); \
     }
 

 float ff_aac_pow34sf_tab[428];
 

 static const uint8_t swb_size_1024_96[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
     64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
 };
 
 static const uint8_t swb_size_1024_64[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
     12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
     40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
 };
 
 static const uint8_t swb_size_1024_48[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
     32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
     96
 };
 
 static const uint8_t swb_size_1024_32[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
     32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
 };
 
 static const uint8_t swb_size_1024_24[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
     32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
 };
 
 static const uint8_t swb_size_1024_16[] = {
     8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
     32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
 };
 
 static const uint8_t swb_size_1024_8[] = {
     12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
     16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
     32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
 };
 

 static const uint8_t *swb_size_1024[] = {

     swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
     swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
     swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
     swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
 };
 
 static const uint8_t swb_size_128_96[] = {
     4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
 };
 
 static const uint8_t swb_size_128_48[] = {
     4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
 };
 
 static const uint8_t swb_size_128_24[] = {
     4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
 };
 
 static const uint8_t swb_size_128_16[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
 };
 
 static const uint8_t swb_size_128_8[] = {
     4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
 };
 

 static const uint8_t *swb_size_128[] = {

     /* the last entry on the following row is swb_size_128_64 but is a
        duplicate of swb_size_128_96 */
     swb_size_128_96, swb_size_128_96, swb_size_128_96,
     swb_size_128_48, swb_size_128_48, swb_size_128_48,
     swb_size_128_24, swb_size_128_24, swb_size_128_16,
     swb_size_128_16, swb_size_128_16, swb_size_128_8
 };
 
 /** default channel configurations */
 static const uint8_t aac_chan_configs[6][5] = {

  {1, TYPE_SCE},                               // 1 channel  - single channel element
  {1, TYPE_CPE},                               // 2 channels - channel pair
  {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
  {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
  {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
  {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE

 };
 
 /**

  * Table to remap channels from libavcodec's default order to AAC order.

  */
 static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
     { 0 },
     { 0, 1 },
     { 2, 0, 1 },
     { 2, 0, 1, 3 },
     { 2, 0, 1, 3, 4 },
     { 2, 0, 1, 4, 5, 3 },
 };
 
 /**

  * Make AAC audio config object.
  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
  */
 static void put_audio_specific_config(AVCodecContext *avctx)
 {
     PutBitContext pb;
     AACEncContext *s = avctx->priv_data;
 
     init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
     put_bits(&pb, 5, 2); //object type - AAC-LC
     put_bits(&pb, 4, s->samplerate_index); //sample rate index

     put_bits(&pb, 4, s->channels);

     //GASpecificConfig
     put_bits(&pb, 1, 0); //frame length - 1024 samples
     put_bits(&pb, 1, 0); //does not depend on core coder
     put_bits(&pb, 1, 0); //is not extension

 
     //Explicitly Mark SBR absent

     put_bits(&pb, 11, 0x2b7); //sync extension

     put_bits(&pb, 5,  AOT_SBR);
     put_bits(&pb, 1,  0);

     flush_put_bits(&pb);
 }
 

 #define WINDOW_FUNC(type) \

 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \

                                     SingleChannelElement *sce, \
                                     const float *audio)

 WINDOW_FUNC(only_long)
 {
     const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;

     float *out = sce->ret_buf;

     fdsp->vector_fmul        (out,        audio,        lwindow, 1024);
     fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);

 }

 WINDOW_FUNC(long_start)
 {
     const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;

     float *out = sce->ret_buf;

     fdsp->vector_fmul(out, audio, lwindow, 1024);

     memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);

     fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);

     memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
 }

 WINDOW_FUNC(long_stop)
 {
     const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;

     float *out = sce->ret_buf;

 
     memset(out, 0, sizeof(out[0]) * 448);

     fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);

     memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);

     fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);

 }

 WINDOW_FUNC(eight_short)
 {
     const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
     const float *in = audio + 448;

     float *out = sce->ret_buf;

     int w;

     for (w = 0; w < 8; w++) {

         fdsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);

         out += 128;
         in  += 128;

         fdsp->vector_fmul_reverse(out, in, swindow, 128);

         out += 128;
     }

 }
 

 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,

                                      SingleChannelElement *sce,
                                      const float *audio) = {

     [ONLY_LONG_SEQUENCE]   = apply_only_long_window,
     [LONG_START_SEQUENCE]  = apply_long_start_window,
     [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
     [LONG_STOP_SEQUENCE]   = apply_long_stop_window
 };
 

 static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
                                   float *audio)

 {

     int i;

     float *output = sce->ret_buf;

     apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);

 
     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)

         s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);

     else
         for (i = 0; i < 1024; i += 128)
             s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
     memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);

 }
 

 /**
  * Encode ics_info element.
  * @see Table 4.6 (syntax of ics_info)
  */

 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)

 {

     int w;

 
     put_bits(&s->pb, 1, 0);                // ics_reserved bit
     put_bits(&s->pb, 2, info->window_sequence[0]);
     put_bits(&s->pb, 1, info->use_kb_window[0]);

     if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {

         put_bits(&s->pb, 6, info->max_sfb);
         put_bits(&s->pb, 1, 0);            // no prediction

     } else {

         put_bits(&s->pb, 4, info->max_sfb);

         for (w = 1; w < 8; w++)

             put_bits(&s->pb, 1, !info->group_len[w]);

     }
 }
 
 /**

  * Encode MS data.
  * @see 4.6.8.1 "Joint Coding - M/S Stereo"

  */

 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)

 {
     int i, w;

 
     put_bits(pb, 2, cpe->ms_mode);

     if (cpe->ms_mode == 1)
         for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])

             for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)

                 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
 }
 
 /**
  * Produce integer coefficients from scalefactors provided by the model.
  */

 static void adjust_frame_information(ChannelElement *cpe, int chans)

 {
     int i, w, w2, g, ch;

     int start, maxsfb, cmaxsfb;

     for (ch = 0; ch < chans; ch++) {

         IndividualChannelStream *ics = &cpe->ch[ch].ics;
         start = 0;
         maxsfb = 0;
         cpe->ch[ch].pulse.num_pulse = 0;

         for (w = 0; w < ics->num_windows*16; w += 16) {
             for (g = 0; g < ics->num_swb; g++) {

                 //apply M/S

                 if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {

                     for (i = 0; i < ics->swb_sizes[g]; i++) {

                         cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
                         cpe->ch[1].coeffs[start+i] =  cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
                     }
                 }
                 start += ics->swb_sizes[g];
             }

             for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
                 ;

             maxsfb = FFMAX(maxsfb, cmaxsfb);
         }
         ics->max_sfb = maxsfb;
 
         //adjust zero bands for window groups

         for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
             for (g = 0; g < ics->max_sfb; g++) {

                 i = 1;

                 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
                     if (!cpe->ch[ch].zeroes[w2*16 + g]) {

                         i = 0;
                         break;
                     }
                 }
                 cpe->ch[ch].zeroes[w*16 + g] = i;
             }
         }
     }
 

     if (chans > 1 && cpe->common_window) {

         IndividualChannelStream *ics0 = &cpe->ch[0].ics;
         IndividualChannelStream *ics1 = &cpe->ch[1].ics;
         int msc = 0;
         ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
         ics1->max_sfb = ics0->max_sfb;

         for (w = 0; w < ics0->num_windows*16; w += 16)
             for (i = 0; i < ics0->max_sfb; i++)

                 if (cpe->ms_mask[w+i])
                     msc++;

         if (msc == 0 || ics0->max_sfb == 0)
             cpe->ms_mode = 0;
         else

             cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;

     }
 }
 
 /**
  * Encode scalefactor band coding type.
  */
 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
 {
     int w;
 

     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])

         s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
 }
 
 /**
  * Encode scalefactors.
  */

 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
                                  SingleChannelElement *sce)

 {
     int off = sce->sf_idx[0], diff;
     int i, w;
 

     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
         for (i = 0; i < sce->ics.max_sfb; i++) {
             if (!sce->zeroes[w*16 + i]) {

                 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;

                 av_assert0(diff >= 0 && diff <= 120);

                 off = sce->sf_idx[w*16 + i];
                 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
             }

         }
     }
 }
 
 /**

  * Encode pulse data.
  */

 static void encode_pulses(AACEncContext *s, Pulse *pulse)

 {
     int i;
 
     put_bits(&s->pb, 1, !!pulse->num_pulse);

     if (!pulse->num_pulse)
         return;

 
     put_bits(&s->pb, 2, pulse->num_pulse - 1);
     put_bits(&s->pb, 6, pulse->start);

     for (i = 0; i < pulse->num_pulse; i++) {

         put_bits(&s->pb, 5, pulse->pos[i]);

         put_bits(&s->pb, 4, pulse->amp[i]);
     }
 }
 
 /**
  * Encode spectral coefficients processed by psychoacoustic model.
  */

 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)

 {

     int start, i, w, w2;

     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {

         start = 0;

         for (i = 0; i < sce->ics.max_sfb; i++) {
             if (sce->zeroes[w*16 + i]) {

                 start += sce->ics.swb_sizes[i];

                 continue;
             }

             for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)

                 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,

                                                    sce->ics.swb_sizes[i],
                                                    sce->sf_idx[w*16 + i],
                                                    sce->band_type[w*16 + i],
                                                    s->lambda);

             start += sce->ics.swb_sizes[i];

         }
     }
 }
 
 /**

  * Encode one channel of audio data.
  */

 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
                                      SingleChannelElement *sce,
                                      int common_window)

 {
     put_bits(&s->pb, 8, sce->sf_idx[0]);

     if (!common_window)
         put_ics_info(s, &sce->ics);

     encode_band_info(s, sce);
     encode_scale_factors(avctx, s, sce);
     encode_pulses(s, &sce->pulse);
     put_bits(&s->pb, 1, 0); //tns
     put_bits(&s->pb, 1, 0); //ssr
     encode_spectral_coeffs(s, sce);
     return 0;
 }
 
 /**

  * Write some auxiliary information about the created AAC file.
  */

 static void put_bitstream_info(AACEncContext *s, const char *name)

 {
     int i, namelen, padbits;
 
     namelen = strlen(name) + 2;

     put_bits(&s->pb, 3, TYPE_FIL);

     put_bits(&s->pb, 4, FFMIN(namelen, 15));

     if (namelen >= 15)

         put_bits(&s->pb, 8, namelen - 14);

     put_bits(&s->pb, 4, 0); //extension type - filler

     padbits = -put_bits_count(&s->pb) & 7;

     avpriv_align_put_bits(&s->pb);

     for (i = 0; i < namelen - 2; i++)

         put_bits(&s->pb, 8, name[i]);
     put_bits(&s->pb, 12 - padbits, 0);
 }
 

 /*

  * Copy input samples.

  * Channels are reordered from libavcodec's default order to AAC order.

  */

 static void copy_input_samples(AACEncContext *s, const AVFrame *frame)

 {

     int ch;
     int end = 2048 + (frame ? frame->nb_samples : 0);
     const uint8_t *channel_map = aac_chan_maps[s->channels - 1];

     /* copy and remap input samples */
     for (ch = 0; ch < s->channels; ch++) {

         /* copy last 1024 samples of previous frame to the start of the current frame */

         memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));

         /* copy new samples and zero any remaining samples */

         if (frame) {

             memcpy(&s->planar_samples[ch][2048],
                    frame->extended_data[channel_map[ch]],
                    frame->nb_samples * sizeof(s->planar_samples[0][0]));

         }

         memset(&s->planar_samples[ch][end], 0,
                (3072 - end) * sizeof(s->planar_samples[0][0]));

     }
 }
 

 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                             const AVFrame *frame, int *got_packet_ptr)

 {
     AACEncContext *s = avctx->priv_data;

     float **samples = s->planar_samples, *samples2, *la, *overlap;

     ChannelElement *cpe;

     int i, ch, w, g, chans, tag, start_ch, ret;

     int chan_el_counter[4];

     FFPsyWindowInfo windows[AAC_MAX_CHANNELS];

     if (s->last_frame == 2)

         return 0;

     /* add current frame to queue */
     if (frame) {

         if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)

             return ret;
     }
 

     copy_input_samples(s, frame);

     if (s->psypp)
         ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);

 
     if (!avctx->frame_number)

         return 0;
 
     start_ch = 0;

     for (i = 0; i < s->chan_map[0]; i++) {

         FFPsyWindowInfo* wi = windows + start_ch;

         tag      = s->chan_map[i+1];

         chans    = tag == TYPE_CPE ? 2 : 1;
         cpe      = &s->cpe[i];

         for (ch = 0; ch < chans; ch++) {
             IndividualChannelStream *ics = &cpe->ch[ch].ics;
             int cur_channel = start_ch + ch;

             overlap  = &samples[cur_channel][0];
             samples2 = overlap + 1024;

             la       = samples2 + (448+64);

             if (!frame)

                 la = NULL;

             if (tag == TYPE_LFE) {

                 wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
                 wi[ch].window_shape   = 0;
                 wi[ch].num_windows    = 1;
                 wi[ch].grouping[0]    = 1;

 
                 /* Only the lowest 12 coefficients are used in a LFE channel.
                  * The expression below results in only the bottom 8 coefficients
                  * being used for 11.025kHz to 16kHz sample rates.
                  */
                 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;

             } else {

                 wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,

                                               ics->window_sequence[0]);

             }

             ics->window_sequence[1] = ics->window_sequence[0];

             ics->window_sequence[0] = wi[ch].window_type[0];

             ics->use_kb_window[1]   = ics->use_kb_window[0];

             ics->use_kb_window[0]   = wi[ch].window_shape;
             ics->num_windows        = wi[ch].num_windows;

             ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];

             ics->num_swb            = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];

             for (w = 0; w < ics->num_windows; w++)
                 ics->group_len[w] = wi[ch].grouping[w];

             apply_window_and_mdct(s, &cpe->ch[ch], overlap);

         }
         start_ch += chans;
     }

     if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)

         return ret;

     do {
         int frame_bits;

 
         init_put_bits(&s->pb, avpkt->data, avpkt->size);
 

         if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))

             put_bitstream_info(s, LIBAVCODEC_IDENT);

         start_ch = 0;
         memset(chan_el_counter, 0, sizeof(chan_el_counter));

         for (i = 0; i < s->chan_map[0]; i++) {

             FFPsyWindowInfo* wi = windows + start_ch;

             const float *coeffs[2];

             tag      = s->chan_map[i+1];

             chans    = tag == TYPE_CPE ? 2 : 1;
             cpe      = &s->cpe[i];

             put_bits(&s->pb, 3, tag);
             put_bits(&s->pb, 4, chan_el_counter[tag]++);

             for (ch = 0; ch < chans; ch++)
                 coeffs[ch] = cpe->ch[ch].coeffs;

             s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);

             for (ch = 0; ch < chans; ch++) {

                 s->cur_channel = start_ch + ch;

                 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);

             }
             cpe->common_window = 0;
             if (chans > 1
                 && wi[0].window_type[0] == wi[1].window_type[0]
                 && wi[0].window_shape   == wi[1].window_shape) {
 
                 cpe->common_window = 1;

                 for (w = 0; w < wi[0].num_windows; w++) {
                     if (wi[0].grouping[w] != wi[1].grouping[w]) {

                         cpe->common_window = 0;
                         break;
                     }

                 }
             }

             s->cur_channel = start_ch;

             if (s->options.stereo_mode && cpe->common_window) {
                 if (s->options.stereo_mode > 0) {
                     IndividualChannelStream *ics = &cpe->ch[0].ics;
                     for (w = 0; w < ics->num_windows; w += ics->group_len[w])
                         for (g = 0;  g < ics->num_swb; g++)
                             cpe->ms_mask[w*16+g] = 1;
                 } else if (s->coder->search_for_ms) {
                     s->coder->search_for_ms(s, cpe, s->lambda);
                 }
             }

             adjust_frame_information(cpe, chans);

             if (chans == 2) {
                 put_bits(&s->pb, 1, cpe->common_window);
                 if (cpe->common_window) {
                     put_ics_info(s, &cpe->ch[0].ics);
                     encode_ms_info(&s->pb, cpe);
                 }

             }

             for (ch = 0; ch < chans; ch++) {
                 s->cur_channel = start_ch + ch;
                 encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);

             }
             start_ch += chans;

         }
 

         frame_bits = put_bits_count(&s->pb);

         if (frame_bits <= 6144 * s->channels - 3) {
             s->psy.bitres.bits = frame_bits / s->channels;

             break;

         }

 
         s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
 
     } while (1);
 

     put_bits(&s->pb, 3, TYPE_END);
     flush_put_bits(&s->pb);
     avctx->frame_bits = put_bits_count(&s->pb);
 
     // rate control stuff

     if (!(avctx->flags & CODEC_FLAG_QSCALE)) {

         float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
         s->lambda *= ratio;

         s->lambda = FFMIN(s->lambda, 65536.f);

     }
 

     if (!frame)

         s->last_frame++;

     ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
                        &avpkt->duration);
 
     avpkt->size = put_bits_count(&s->pb) >> 3;
     *got_packet_ptr = 1;
     return 0;

 }
 

 static av_cold int aac_encode_end(AVCodecContext *avctx)
 {
     AACEncContext *s = avctx->priv_data;
 
     ff_mdct_end(&s->mdct1024);
     ff_mdct_end(&s->mdct128);

     ff_psy_end(&s->psy);

     if (s->psypp)
         ff_psy_preprocess_end(s->psypp);

     av_freep(&s->buffer.samples);

     av_freep(&s->cpe);

     ff_af_queue_close(&s->afq);

     return 0;
 }
 

 static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 {
     int ret = 0;
 

     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

 
     // window init
     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
     ff_init_ff_sine_windows(10);
     ff_init_ff_sine_windows(7);
 

     if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))

         return ret;

     if (ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0))

         return ret;
 
     return 0;
 }
 
 static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
 {

     int ch;

     FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);

     FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
     FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
 

     for(ch = 0; ch < s->channels; ch++)

         s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;

     return 0;
 alloc_fail:
     return AVERROR(ENOMEM);
 }
 
 static av_cold int aac_encode_init(AVCodecContext *avctx)
 {
     AACEncContext *s = avctx->priv_data;
     int i, ret = 0;
     const uint8_t *sizes[2];
     uint8_t grouping[AAC_MAX_CHANNELS];
     int lengths[2];
 
     avctx->frame_size = 1024;
 
     for (i = 0; i < 16; i++)
         if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
             break;
 

     s->channels = avctx->channels;
 

     ERROR_IF(i == 16,
              "Unsupported sample rate %d\n", avctx->sample_rate);

     ERROR_IF(s->channels > AAC_MAX_CHANNELS,
              "Unsupported number of channels: %d\n", s->channels);

     ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
              "Unsupported profile %d\n", avctx->profile);

     ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,

              "Too many bits per frame requested\n");
 
     s->samplerate_index = i;
 

     s->chan_map = aac_chan_configs[s->channels-1];

 
     if (ret = dsp_init(avctx, s))
         goto fail;
 
     if (ret = alloc_buffers(avctx, s))
         goto fail;
 
     avctx->extradata_size = 5;
     put_audio_specific_config(avctx);
 
     sizes[0]   = swb_size_1024[i];
     sizes[1]   = swb_size_128[i];
     lengths[0] = ff_aac_num_swb_1024[i];
     lengths[1] = ff_aac_num_swb_128[i];
     for (i = 0; i < s->chan_map[0]; i++)
         grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
     if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
         goto fail;
     s->psypp = ff_psy_preprocess_init(avctx);

     s->coder = &ff_aac_coders[s->options.aac_coder];

     if (HAVE_MIPSDSPR1)
         ff_aac_coder_init_mips(s);
 

     s->lambda = avctx->global_quality ? avctx->global_quality : 120;
 
     ff_aac_tableinit();
 

     for (i = 0; i < 428; i++)
         ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
 

     avctx->delay = 1024;
     ff_af_queue_init(avctx, &s->afq);
 

     return 0;
 fail:
     aac_encode_end(avctx);
     return ret;
 }
 

 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption aacenc_options[] = {

     {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},

         {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
         {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
         {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},

     {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = 2}, 0, AAC_CODER_NB-1, AACENC_FLAGS},

     {NULL}
 };
 
 static const AVClass aacenc_class = {
     "AAC encoder",
     av_default_item_name,
     aacenc_options,
     LIBAVUTIL_VERSION_INT,
 };
 

 /* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
  * failures */
 static const int mpeg4audio_sample_rates[16] = {
     96000, 88200, 64000, 48000, 44100, 32000,
     24000, 22050, 16000, 12000, 11025, 8000, 7350
 };
 

 AVCodec ff_aac_encoder = {

     .name           = "aac",
     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_AAC,

     .priv_data_size = sizeof(AACEncContext),
     .init           = aac_encode_init,

     .encode2        = aac_encode_frame,

     .close          = aac_encode_end,

     .supported_samplerates = mpeg4audio_sample_rates,

     .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
                       CODEC_CAP_EXPERIMENTAL,

     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,

                                                      AV_SAMPLE_FMT_NONE },

     .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),

     .priv_class     = &aacenc_class,

 };